Recherche avancée
Autres articles (100)
-
MediaSPIP 0.1 Beta version
25 avril 2011, parMediaSPIP 0.1 beta is the first version of MediaSPIP proclaimed as "usable".
The zip file provided here only contains the sources of MediaSPIP in its standalone version.
To get a working installation, you must manually install all-software dependencies on the server.
If you want to use this archive for an installation in "farm mode", you will also need to proceed to other manual (...) -
Multilang : améliorer l’interface pour les blocs multilingues
18 février 2011, parMultilang est un plugin supplémentaire qui n’est pas activé par défaut lors de l’initialisation de MediaSPIP.
Après son activation, une préconfiguration est mise en place automatiquement par MediaSPIP init permettant à la nouvelle fonctionnalité d’être automatiquement opérationnelle. Il n’est donc pas obligatoire de passer par une étape de configuration pour cela. -
Amélioration de la version de base
13 septembre 2013Jolie sélection multiple
Le plugin Chosen permet d’améliorer l’ergonomie des champs de sélection multiple. Voir les deux images suivantes pour comparer.
Il suffit pour cela d’activer le plugin Chosen (Configuration générale du site > Gestion des plugins), puis de configurer le plugin (Les squelettes > Chosen) en activant l’utilisation de Chosen dans le site public et en spécifiant les éléments de formulaires à améliorer, par exemple select[multiple] pour les listes à sélection multiple (...)
Sur d’autres sites (11828)
-
keepalive type and frequency in ffmpeg [on hold]
19 novembre 2013, par Jack SimthMy company has a bunch of IP cameras that we distribute - specifically Grandstream - and the manufacturer has changed their firmware. The normal keepalive that ffmpeg uses for the rtsp streams ( either ff_rtsp_send_cmd_async(s, "GET_PARAMETER", rt->control_uri, NULL) ; or ff_rtsp_send_cmd_async(s, "OPTIONS", "*", NULL) ; both in in libavformat/rtspdec.c) is no longer working, for two reasons :
1) The new Grandstream firmware is now checking for a receiver report to determine whether or not the program reading the stream is live, not just anything.
2) The new Grandstream firmware requires that the receiver report to keep the connection alive happen at least once every 25 seconds, and on the audio stream it is currently only happening about every 30 seconds or so (video is getting it every 7 seconds or so).
So after about a minute with ffmpeg connected, the camera stops sending the audio stream, the audio stream on ffmpeg reads end-of-file, and then ffmpeg shuts everything down.
As I can't change the firmware, I'm trying to dig through the ffmpeg code to make it send the appropriate receiver report for the keep alive... but I am getting nowhere. I've added a little snippet of code into the receiver reports so I know when they're running when I call ffmpeg on debug, but... well, it's not going well.
Test command :
ffmpeg -loglevel debug -i rtsp ://admin:admin@192.168.4.3:554/0 -acodec libmp3lame -ar 22050 -vcodec copy -y -f flv /dev/null &> test.txtTest output :
`[root@localhost ffmpeg]# cat test.txt
ffmpeg version 2.0 Copyright (c) 2000-2013 the FFmpeg developers
built on Aug 21 2013 14:24:28 with gcc 4.4.7 (GCC) 20120313 (Red Hat 4.4.7-3)
configuration: --datadir=/usr/share/ffmpeg --bindir=/usr/local/bin --libdir=/usr/local/lib --incdir=/usr/local/include --shlibdir=/usr/lib --mandir=/usr/share/man --disable-avisynth --extra-cflags='-O2 -g -pipe -Wall -Wp,-D_FORTIFY_SOURCE=2 -fexceptions -fstack-protector --param=ssp-buffer-size=4 -m32 -march=i386 -mtune=generic -fasynchronous-unwind-tables' --enable-avfilter --enable-libx264 --enable-gpl --enable-version3 --enable-postproc --enable-pthreads --enable-shared --enable-swscale --enable-vdpau --enable-x11grab --enable-librtmp --enable-libopencore-amrnb --enable-libopencore-amrwb --disable-static --enable-libgsm --enable-libxvid --enable-libvpx --enable-libvorbis --enable-libvo-aacenc --enable-libmp3lame
libavutil 52. 38.100 / 52. 38.100
libavcodec 55. 18.102 / 55. 18.102
libavformat 55. 12.100 / 55. 12.100
libavdevice 55. 3.100 / 55. 3.100
libavfilter 3. 79.101 / 3. 79.101
libswscale 2. 3.100 / 2. 3.100
libswresample 0. 17.102 / 0. 17.102
libpostproc 52. 3.100 / 52. 3.100
Splitting the commandline.
Reading option '-loglevel' ... matched as option 'loglevel' (set logging level) with argument 'debug'.
Reading option '-i' ... matched as input file with argument 'rtsp://admin:admin@192.168.4.3:554/0'.
Reading option '-acodec' ... matched as option 'acodec' (force audio codec ('copy' to copy stream)) with argument 'libmp3lame'.
Reading option '-ar' ... matched as option 'ar' (set audio sampling rate (in Hz)) with argument '22050'.
Reading option '-vcodec' ... matched as option 'vcodec' (force video codec ('copy' to copy stream)) with argument 'copy'.
Reading option '-y' ... matched as option 'y' (overwrite output files) with argument '1'.
Reading option '-f' ... matched as option 'f' (force format) with argument 'flv'.
Reading option '/dev/null' ... matched as output file.
Finished splitting the commandline.
Parsing a group of options: global .
Applying option loglevel (set logging level) with argument debug.
Applying option y (overwrite output files) with argument 1.
Successfully parsed a group of options.
Parsing a group of options: input file rtsp://admin:admin@192.168.4.3:554/0.
Successfully parsed a group of options.
Opening an input file: rtsp://admin:admin@192.168.4.3:554/0.
[rtsp @ 0x9d9ccc0] SDP:
v=0
o=StreamingServer 3331435948 1116907222000 IN IP4 192.168.4.3
s=h264.mp4
c=IN IP4 0.0.0.0
t=0 0
a=control:*
m=video 0 RTP/AVP 96
a=control:trackID=0
a=rtpmap:96 H264/90000
a=fmtp:96 packetization-mode=1; sprop-parameter-sets=Z0LgHtoCgPRA,aM4wpIA=
m=audio 0 RTP/AVP 0
a=control:trackID=1
a=rtpmap:0 PCMU/8000
a=ptime:20
m=application 0 RTP/AVP 107
a=control:trackID=2
a=rtpmap:107 vnd.onvif.metadata/90000
[rtsp @ 0x9d9ccc0] video codec set to: h264
[NULL @ 0x9d9f400] RTP Packetization Mode: 1
[NULL @ 0x9d9f400] Extradata set to 0x9d9f900 (size: 22)!
[rtsp @ 0x9d9ccc0] audio codec set to: pcm_mulaw
[rtsp @ 0x9d9ccc0] audio samplerate set to: 8000
[rtsp @ 0x9d9ccc0] audio channels set to: 1
[rtsp @ 0x9d9ccc0] hello state=0
[h264 @ 0x9d9f400] Current profile doesn't provide more RBSP data in PPS, skipping
Last message repeated 1 times
[rtsp @ 0x9d9ccc0] All info found
Guessed Channel Layout for Input Stream #0.1 : mono
Input #0, rtsp, from 'rtsp://admin:admin@192.168.4.3:554/0':
Metadata:
title : h264.mp4
Duration: N/A, start: 0.000000, bitrate: 64 kb/s
Stream #0:0, 28, 1/90000: Video: h264 (Constrained Baseline), yuv420p, 640x480, 1/180000, 10 tbr, 90k tbn, 180k tbc
Stream #0:1, 156, 1/8000: Audio: pcm_mulaw, 8000 Hz, mono, s16, 64 kb/s
Successfully opened the file.
Parsing a group of options: output file /dev/null.
Applying option acodec (force audio codec ('copy' to copy stream)) with argument libmp3lame.
Applying option ar (set audio sampling rate (in Hz)) with argument 22050.
Applying option vcodec (force video codec ('copy' to copy stream)) with argument copy.
Applying option f (force format) with argument flv.
Successfully parsed a group of options.
Opening an output file: /dev/null.
Successfully opened the file.
detected 2 logical cores
[graph 0 input from stream 0:1 @ 0x9f15380] Setting 'time_base' to value '1/8000'
[graph 0 input from stream 0:1 @ 0x9f15380] Setting 'sample_rate' to value '8000'
[graph 0 input from stream 0:1 @ 0x9f15380] Setting 'sample_fmt' to value 's16'
[graph 0 input from stream 0:1 @ 0x9f15380] Setting 'channel_layout' to value '0x4'
[graph 0 input from stream 0:1 @ 0x9f15380] tb:1/8000 samplefmt:s16 samplerate:8000 chlayout:0x4
[audio format for output stream 0:1 @ 0x9efa7c0] Setting 'sample_fmts' to value 's32p|fltp|s16p'
[audio format for output stream 0:1 @ 0x9efa7c0] Setting 'sample_rates' to value '22050'
[audio format for output stream 0:1 @ 0x9efa7c0] Setting 'channel_layouts' to value '0x4|0x3'
[audio format for output stream 0:1 @ 0x9efa7c0] auto-inserting filter 'auto-inserted resampler 0' between the filter 'Parsed_anull_0' and the filter 'audio format for output stream 0:1'
[AVFilterGraph @ 0x9f15980] query_formats: 4 queried, 9 merged, 3 already done, 0 delayed
[auto-inserted resampler 0 @ 0x9dfada0] ch:1 chl:mono fmt:s16 r:8000Hz -> ch:1 chl:mono fmt:s16p r:22050Hz
Output #0, flv, to '/dev/null':
Metadata:
title : h264.mp4
encoder : Lavf55.12.100
Stream #0:0, 0, 1/1000: Video: h264 ([7][0][0][0] / 0x0007), yuv420p, 640x480, 1/90000, q=2-31, 1k tbn, 90k tbc
Stream #0:1, 0, 1/1000: Audio: mp3 (libmp3lame) ([2][0][0][0] / 0x0002), 22050 Hz, mono, s16p
Stream mapping:
Stream #0:0 -> #0:0 (copy)
Stream #0:1 -> #0:1 (pcm_mulaw -> libmp3lame)
Press [q] to stop, [?] for help
Current profile doesn't provide more RBSP data in PPS, skippingrate= 135.4kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 134.4kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 135.0kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 135.5kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 136.9kbits/s
Queue input is backward in time= 233kB time=00:00:13.69 bitrate= 139.4kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 136.3kbits/s
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 13926; changing to 14239. This may result in incorrect timestamps in the output file.
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 13952; changing to 14239. This may result in incorrect timestamps in the output file.
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 13979; changing to 14239. This may result in incorrect timestamps in the output file.
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14005; changing to 14239. This may result in incorrect timestamps in the output file.
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14031; changing to 14239. This may result in incorrect timestamps in the output file.
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14057; changing to 14239. This may result in incorrect timestamps in the output file.
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14083; changing to 14239. This may result in incorrect timestamps in the output file.
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14109; changing to 14239. This may result in incorrect timestamps in the output file.
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14135; changing to 14239. This may result in incorrect timestamps in the output file.
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14161; changing to 14239. This may result in incorrect timestamps in the output file.
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14188; changing to 14239. This may result in incorrect timestamps in the output file.
[flv @ 0x9de1200] Non-monotonous DTS in output stream 0:1; previous: 14239, current: 14214; changing to 14239. This may result in incorrect timestamps in the output file.
Current profile doesn't provide more RBSP data in PPS, skippingrate= 141.5kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 142.0kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 142.5kbits/s
Receiver Report delay: 469789, gettime: -1527669086, last_recep: 322446, timebase: -1534837492
Current profile doesn't provide more RBSP data in PPS, skippingrate= 141.5kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 141.7kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 141.1kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 140.6kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 140.7kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.9kbits/s
Receiver Report delay: 132993, gettime: -1516538925, last_recep: 322446, timebase: -1518568234
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.6kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.6kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.7kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.4kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 140.0kbits/s
Receiver Report delay: 897727, gettime: -1504870331, last_recep: 322446, timebase: -1518568552
[NULL @ 0x9d9f400] Current profile doesn't provide more RBSP data in PPS, skipping
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.4kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.1kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.0kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 139.0kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 138.6kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 138.5kbits/s
Current profile doesn't provide more RBSP data in PPS, skippingrate= 138.2kbits/s
EOF on sink link output stream 0:1:default.time=00:00:58.40 bitrate= 139.6kbits/s
No more output streams to write to, finishing.
[libmp3lame @ 0x9dfa580] Trying to remove 344 more samples than there are in the queue
frame= 589 fps= 11 q=-1.0 Lsize= 1003kB time=00:00:58.85 bitrate= 139.5kbits/s
video:724kB audio:231kB subtitle:0 global headers:0kB muxing overhead 4.955356%
2959 frames successfully decoded, 0 decoding errors
[AVIOContext @ 0x9e021c0] Statistics: 3 seeks, 2860 writeouts
[root@localhost ffmpeg]# -
FFmpeg playback on iOS out of sync after conversion
26 novembre 2013, par YYZSo I've got a mov file, and I'm converting it to .mp4 and then playing it back on both iOS and Android. There are no sync problems on Android, but iOS videos are always slightly out of sync.
Here is my ffmpeg command :
ffmpeg -i original.mov -threads 4 -c:a aac -cutoff 15000 -ab 96k -ar 22050 -c:v libx264 -s 638x360 -profile:v baseline -r 25 -movflags faststart -map_metadata:s:v -1:g -strict experimental -y out.mp4Is there anything obvious that stands out as wrong ?
-
FFMPEG wmv conversion to flv
25 novembre 2012, par Brandon Grossuttianyone using ffmpeg
I have a fairly simple wmv exported by a user from movie maker with standard output and want to convert to .flv using
C :>ffmpeg -i "E :\Jab Core 4 Recounters.wmv" -vcodec flv "C :\Net Projects\SVN\IntegratedAlgorithmics\src\MediaAdmin\MediaAdmin\bin\Debug\Movies\Jab Core 4 Recounters.flv" -ar 44100
the output / error i receive is
FFmpeg version 0.5, Copyright (c) 2000-2009 Fabrice Bellard, et al.
configuration: --enable-gpl --enable-postproc --enable-swscale --enable-avfilt
er --enable-avfilter-lavf --enable-pthreads --enable-avisynth --enable-libfaac -
-enable-libfaad --enable-libmp3lame --enable-libspeex --enable-libtheora --enabl
e-libvorbis --enable-libxvid --enable-libx264 --enable-memalign-hack
libavutil 49.15. 0 / 49.15. 0
libavcodec 52.20. 0 / 52.20. 0
libavformat 52.31. 0 / 52.31. 0
libavdevice 52. 1. 0 / 52. 1. 0
libavfilter 0. 4. 0 / 0. 4. 0
libswscale 0. 7. 1 / 0. 7. 1
libpostproc 51. 2. 0 / 51. 2. 0
built on Mar 16 2009 16:09:18, gcc: 4.2.4 [Sherpya]
[wmv3 @ 0x1c0d490]Extra data: 8 bits left, value: 0
Seems stream 1 codec frame rate differs from container frame rate: 1000.00 (1000
/1) -> 30.00 (30/1)
Input #0, asf, from 'E:\Jab Core 4 Recounters.wmv':
Duration: 00:01:55.99, start: 5.000000, bitrate: 813 kb/s
Stream #0.0: Audio: wmav2, 48000 Hz, stereo, s16, 192 kb/s
Stream #0.1: Video: wmv3, yuv420p, 640x480, 586 kb/s, 30 tbr, 1k tbn, 1k tbc
Output #0, flv, to 'C:\Net Projects\SVN\IntegratedAlgorithmics\src\MediaAdmin\Me
diaAdmin\bin\Debug\Movies\Jab Core 4 Recounters.flv':
Stream #0.0: Video: flv, yuv420p, 640x480, q=2-31, 200 kb/s, 90k tbn, 30 tbc
Stream #0.1: Audio: libmp3lame, 48000 Hz, stereo, s16, 64 kb/s
Stream mapping:
Stream #0.1 -> #0.0
Stream #0.0 -> #0.1
[wmv3 @ 0x1c0d490]Extra data: 8 bits left, value: 0
[libmp3lame @ 0x1c0d8d0]flv does not support that sample rate, choose from (4410
0, 22050, 11025).
Could not write header for output file #0 (incorrect codec parameters ?)i added th -ar switch when i got the error the first time
the codec info i have on the file is as follows
General
Complete name : E:\Jab Core 4 Recounters.wmv
Format : Windows Media
File size : 11.3 MiB
Duration : 2mn 0s
Overall bit rate mode : Variable
Overall bit rate : 780 Kbps
Maximum Overall bit rate : 949 Kbps
Encoded date : UTC 2009-03-07 07:02:41.121
Writing application : 6.0.6000.16386 / Windows Movie Maker
Application : Windows Movie Maker 6.0.6000.16386
Video
ID : 2
Format : VC-1
Format profile : MP@ML
Codec ID : WMV3
Codec ID/Info : Windows Media Video 9
Codec ID/Hint : WMV3
Duration : 2mn 0s
Bit rate mode : Variable
Bit rate : 587 Kbps
Width : 640 pixels
Height : 480 pixels
Display aspect ratio : 4/3
Frame rate : 30.000 fps
Resolution : 24 bits
Scan type : Progressive
Bits/(Pixel*Frame) : 0.064
Stream size : 8.46 MiB (75%)
Language : en-us
Audio
ID : 1
Format : WMA2
Format profile : L3
Codec ID : 161
Codec ID/Info : Windows Media Audio 2
Description of the codec : Windows Media Audio 9.2 - VBR Quality 90, 48 kHz, stereo 1-pass VBR
Duration : 2mn 0s
Bit rate mode : Variable
Bit rate : 186 Kbps
Channel(s) : 2 channels
Sampling rate : 48.0 KHz
Resolution : 16 bits
Stream size : 2.68 MiB (24%)
Language : en-usi see alot of people with this issue with so solution or cause
any ideas would be helpful
thanks in advance
Syndiquer tout le site
